The multi-functional IP-PBX delivers high performance. Simplified management, reduced communication cost, seamless connectivity with remote users and between geographically dispersed branches, advanced communication means and enhanced productivity are apparent benefits. The system employs open-standard SIP protocol and is hence interoperable with SIP proxies, gateways and IP phones. Communication of small and mid-sized enterprises as well as geographically distributed offices, remote workers and contact centre is much simplified and enhanced with SAPEX.
The Session Initiation Protocol (SIP) is an application-layer signaling protocol used for establishing multimedia sessions over Internet Protocol (IP) networks, such as voice or video calls. SIP can be used to initiate a call, invite participants and manage a call. It defines several methods for call setup, maintenance and termination. Common SIP methods include INVITE for call initiation, ACK to acknowledge call setup, BYE to terminate a call, and REGISTER for registering location. SIP uses SDP for negotiating media capabilities and RTP for transporting media streams.
The document provides an overview of the Genesys SIP Server, including its fundamental purpose, architecture, deployment modes, load balancing capabilities, multi-threaded design, multi-site support, and important network considerations for ensuring quality voice services. The SIP Server combines call switching and T-Server functionality, and can operate with or without a third-party softswitch in various deployment configurations. Proper network sizing, bandwidth provisioning, quality of service controls, and remote access methods are critical to delivering high quality voice.
The Session Initiation Protocol (SIP) is the dominant signaling protocol used in VoIP today. It is
responsible for the establishment, control and termination of sessions by exchanging ASCII-text-based
messages between the endpoints. This post goes through the basic components of SIP: messages and
logical entities.
Session Initiation Protocol is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
Brief introduction into SIP protocol, how it works, common problems to solve. Tech. details about handshake, SIP Trunks and SIP trunking. Market research.
This document provides an overview of three common voice over IP protocols: SIP, H.323, and MGCP. It describes the basic architecture and components of SIP, including users agents, proxies, registrars, and how SIP establishes calls. It also summarizes H.323, describing its terminals, gateways, gatekeepers, and call establishment. MGCP is briefly mentioned as another VOIP protocol.
The document provides an overview of three common voice over IP protocols: SIP, H.323, and MGCP. It describes the basic architecture and components of SIP and H.323, how they establish communication sessions, and compares some of their key differences and strengths.
SIP - More than meets the eye
Speakers:
Ofer Cohen - VOIP Group Leader, LivePerson
Yossi Maimon - VOIP Technical Leader, LivePerson
An Introduction to the SIP protocol.
SIP Position in telecommunication networks and the content services.
What is SIP:
The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating sessions consisting of one or several media streams. SIP can be used for two-party (unicast) or multiparty (multicast) sessions. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.
(Source: Wikipedia)
The Session Initiation Protocol (SIP) is an application-layer signaling protocol used for establishing multimedia sessions over Internet Protocol (IP) networks, such as voice or video calls. SIP can be used to initiate a call, invite participants and manage a call. It defines several methods for call setup, maintenance and termination. Common SIP methods include INVITE for call initiation, ACK to acknowledge call setup, BYE to terminate a call, and REGISTER for registering location. SIP uses SDP for negotiating media capabilities and RTP for transporting media streams.
The document provides an overview of the Genesys SIP Server, including its fundamental purpose, architecture, deployment modes, load balancing capabilities, multi-threaded design, multi-site support, and important network considerations for ensuring quality voice services. The SIP Server combines call switching and T-Server functionality, and can operate with or without a third-party softswitch in various deployment configurations. Proper network sizing, bandwidth provisioning, quality of service controls, and remote access methods are critical to delivering high quality voice.
The Session Initiation Protocol (SIP) is the dominant signaling protocol used in VoIP today. It is
responsible for the establishment, control and termination of sessions by exchanging ASCII-text-based
messages between the endpoints. This post goes through the basic components of SIP: messages and
logical entities.
Session Initiation Protocol is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
Brief introduction into SIP protocol, how it works, common problems to solve. Tech. details about handshake, SIP Trunks and SIP trunking. Market research.
This document provides an overview of three common voice over IP protocols: SIP, H.323, and MGCP. It describes the basic architecture and components of SIP, including users agents, proxies, registrars, and how SIP establishes calls. It also summarizes H.323, describing its terminals, gateways, gatekeepers, and call establishment. MGCP is briefly mentioned as another VOIP protocol.
The document provides an overview of three common voice over IP protocols: SIP, H.323, and MGCP. It describes the basic architecture and components of SIP and H.323, how they establish communication sessions, and compares some of their key differences and strengths.
SIP - More than meets the eye
Speakers:
Ofer Cohen - VOIP Group Leader, LivePerson
Yossi Maimon - VOIP Technical Leader, LivePerson
An Introduction to the SIP protocol.
SIP Position in telecommunication networks and the content services.
What is SIP:
The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating sessions consisting of one or several media streams. SIP can be used for two-party (unicast) or multiparty (multicast) sessions. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.
(Source: Wikipedia)
SIP is an application-layer protocol for establishing multimedia sessions over IP networks. It can be used to initiate voice, video, and instant messaging communications. SIP works by having user agents (clients and servers) exchange SIP request and response messages. These messages contain information about session setup, modification, and termination. Some key SIP components include user agents, proxy servers, registrar servers, and redirect servers. SIP messages use a request-response transaction model and contain start lines, headers, and optional message bodies. Common request methods are INVITE, ACK, BYE, and REGISTER. Typical response codes include 100-199 (informational), 200-299 (success), 300-399 (redirection), 400-499
The presentation is a compiled assembly from the SIP RFC' s, and original works of Alan Johnston and Henry Sinnreich . It contains Sip Detailed , Call flows , Architecture descriptions , SIP services , sip security , sip programming.
The document provides an overview of three main voice over IP protocols: SIP, H.323, and MGCP. It describes the architecture and components of SIP and H.323, how they establish calls, and compares some key differences between the two protocols. MGCP is also introduced but not described in detail.
M.Sreenath seeks a position as a Genesys Engineer utilizing over 2 years of experience implementing and maintaining contact centers using Genesys voice routing and reporting. He has experience developing, configuring, and testing large-scale voice solutions, and is familiar with all aspects of technology projects. Sreenath has a B.Tech in Electronics and Communications Engineering from JNTU and certifications in CISCO and Genesys frameworks.
1. The document introduces Session Initiation Protocol (SIP), explaining that it is an application layer signaling protocol for initiating, modifying, and terminating multimedia communication sessions over IP such as voice and video calls.
2. It describes why SIP is used, including for conferencing, distance learning, video conferencing, instant messaging, and voice calls. It also outlines the main components of a SIP network including user agents, proxies, and redirects servers.
3. The document provides an overview of how SIP works by outlining the signaling process for registration, call setup and teardown, redirection, and media routing between user agents.
This document describes a CMPE 208 Fall 2008 project on the Session Initiation Protocol (SIP) carried out by Chinmay Padhye, Amit More, Abhishek Sharma, and Nihar Dandekar. The project introduces SIP, describes its entities and functions, and outlines test cases the group planned to carry out including softphone registration, call setup, call forwarding, and call forking using softphones like Xlite and a Hamachi server.
This document provides an overview of the Session Initiation Protocol (SIP) call flows and compliance information for Cisco SIP gateways. It describes SIP messages, methods, responses and headers. It also includes illustrations and explanations of successful and failed call flow scenarios between SIP gateways and SIP IP phones. Additional appendices provide information on third party call control and SIP diversion header implementation.
SIP is a signaling protocol that establishes, manages, and terminates multimedia sessions over IP networks. It uses components like user agents, registrars, proxies, and redirect servers. A SIP call is established through a three-way handshake between these components. SIP supports features like registration, redirection, forking, and mobility through its architecture and use of protocols like SIP, SDP, RTP, and MGCP.
Introduction to SIP(Session Initiation Protocol)William Lee
Session Initiation Protocol (SIP) is a signaling protocol for managing multimedia communication sessions over Internet Protocol (IP) networks. SIP can be used to establish two-party or multiparty sessions that include voice, video, chat, gaming, and other forms of media. The document introduces SIP architecture, message format, and common call flows including registration, basic call setup, call modification, call hold, and three-way conferencing.
One of the most widely observed fraud scenarios is the case of a malicious user detecting the address of a PSTN gateway and accessing that gateway directly. Once the attacker has managed to access the gateway the attacker can start selling telephony minutes through that
gateway.
The ABC SBC establishes a secure border between the VoIP service provider’s core VoIP components–e.g., PSTN gateway, SIP proxy and application servers- and the subscribers. As the border element, the ABC SBC hides the details of the operator’s network from subscribers
and absorbs any attacks and sudden spikes in the subscriber traffic. Further, the mediation features of the ABC SBC shield the operator’s network from malfunctioning user agents and any interoperability issues.
Interaction Dialer is an outbound dialing solution that can increase revenue, decrease labor costs, and optimize calling campaigns through features like predictive dialing, powerful ACD, and regulatory compliance tools. It provides automated dialing capabilities to improve agent productivity by reducing non-productive tasks. The system uses a single software platform to integrate features like a dialer, recording, reporting, and other functions.
Avaya Aura Application Enablement Services (AES)Motty Ben Atia
The document discusses Avaya's Application Enablement Services (AES) platform. AES provides protocols, APIs, and web services that enable development of modular communication applications. It opens up functionality of Communication Manager and adds communication capabilities to desktop tools. AES supports integration of communication and business applications through API connectors and web services.
This document summarizes hosted VoIP services provided by Reignmaker Communications. It defines VoIP and hosted VoIP, explaining that hosted VoIP provides phone system features over the internet rather than traditional phone lines. It outlines key advantages like lower costs, scalability, and built-in disaster recovery. Finally, it introduces Reignmaker's service offerings and positioning as an end-to-end provider with nationwide coverage and exceptional support.
The document describes the SAPEX IP-PBX server. It is an all-in-one embedded IP-PBX platform that integrates registrar, proxy, presence, and voice mail servers. It supports SIP protocol and allows registration of multiple SIP trunks. Key features include back-to-back user agent for call management, dynamic DNS for seamless connectivity, presence and instant messaging, and RADIUS client for call logging and accounting. The system provides various call routing, call management, and telephony features.
The document describes the features and capabilities of the SAPEX IP-PBX server. It is an all-in-one embedded IP-PBX platform that integrates registrar, proxy, presence and voice mail servers. It supports SIP protocol and allows registration of multiple SIP trunks. Key features include seamless connectivity for remote users, dynamic DNS, presence and instant messaging, call detail recording, RADIUS client for authentication and accounting, and various call management functions.
SPARSH VP248 is a high-definition VoIP phone built with superior acoustics and elegant design to provide unsurpassed audio quality and rich user experience.
Based on open-standard SIP protocol, SPARSH VP248 is interoperable with any standard SIP infrastructure such as IP-PBX, SIP Proxies, Softswitches and Stand-alone applications.
SPARSH VP248 is designed for power users, knowledge workers and managers for quick access totheadvance system features and functions. A feature-packed IP phoneenables user to work efficiently with advance call handling capabilities
Matrix Telecom Solutions: SETU VTEP - Fixed VoIP to T1/E1 PRI GatewayMatrix Comsec
Matrix SETU VTEP is a compact and dedicated gateway for VoIP to T1/E1 PRI network offering high-value communication experience to businesses of all size, Service Providers, Call Centers and simple but cost-effective solution for multi-location branch office communication. This intelligently designed gateway incorporates advanced features with multiple connectivity options to connect with a legacy communication system using T1/E1 or PRI signaling. SETU VTEP offers reliable and cost-effective solutions to the changing requirements of the business communication and offer customer value for money.
The document introduces the Vertical Wave IP 2500, an IP PBX system with integrated applications. It discusses key customer considerations for IP phone systems and the evolution from legacy systems to converged platforms. The Vertical Wave provides scalability, flexibility, investment protection, mobility and ease of management through its applications-inside architecture and web-based administration tools.
Advantage2000ucs version 1-2 InternationalNajam Khan
The document describes the features of Advantage 2000 Unified Communication Server (UCS), a communication platform that provides unified communications, collaboration, and integration capabilities. It includes features such as IP PBX, call center software, CRM software, instant messaging, video calling, conferencing, and supports integration with smartphones, softphones, and third-party software. The UCS provides these capabilities to small and medium businesses through an affordable and easy to manage solution.
SIP is an application-layer protocol for establishing multimedia sessions over IP networks. It can be used to initiate voice, video, and instant messaging communications. SIP works by having user agents (clients and servers) exchange SIP request and response messages. These messages contain information about session setup, modification, and termination. Some key SIP components include user agents, proxy servers, registrar servers, and redirect servers. SIP messages use a request-response transaction model and contain start lines, headers, and optional message bodies. Common request methods are INVITE, ACK, BYE, and REGISTER. Typical response codes include 100-199 (informational), 200-299 (success), 300-399 (redirection), 400-499
The presentation is a compiled assembly from the SIP RFC' s, and original works of Alan Johnston and Henry Sinnreich . It contains Sip Detailed , Call flows , Architecture descriptions , SIP services , sip security , sip programming.
The document provides an overview of three main voice over IP protocols: SIP, H.323, and MGCP. It describes the architecture and components of SIP and H.323, how they establish calls, and compares some key differences between the two protocols. MGCP is also introduced but not described in detail.
M.Sreenath seeks a position as a Genesys Engineer utilizing over 2 years of experience implementing and maintaining contact centers using Genesys voice routing and reporting. He has experience developing, configuring, and testing large-scale voice solutions, and is familiar with all aspects of technology projects. Sreenath has a B.Tech in Electronics and Communications Engineering from JNTU and certifications in CISCO and Genesys frameworks.
1. The document introduces Session Initiation Protocol (SIP), explaining that it is an application layer signaling protocol for initiating, modifying, and terminating multimedia communication sessions over IP such as voice and video calls.
2. It describes why SIP is used, including for conferencing, distance learning, video conferencing, instant messaging, and voice calls. It also outlines the main components of a SIP network including user agents, proxies, and redirects servers.
3. The document provides an overview of how SIP works by outlining the signaling process for registration, call setup and teardown, redirection, and media routing between user agents.
This document describes a CMPE 208 Fall 2008 project on the Session Initiation Protocol (SIP) carried out by Chinmay Padhye, Amit More, Abhishek Sharma, and Nihar Dandekar. The project introduces SIP, describes its entities and functions, and outlines test cases the group planned to carry out including softphone registration, call setup, call forwarding, and call forking using softphones like Xlite and a Hamachi server.
This document provides an overview of the Session Initiation Protocol (SIP) call flows and compliance information for Cisco SIP gateways. It describes SIP messages, methods, responses and headers. It also includes illustrations and explanations of successful and failed call flow scenarios between SIP gateways and SIP IP phones. Additional appendices provide information on third party call control and SIP diversion header implementation.
SIP is a signaling protocol that establishes, manages, and terminates multimedia sessions over IP networks. It uses components like user agents, registrars, proxies, and redirect servers. A SIP call is established through a three-way handshake between these components. SIP supports features like registration, redirection, forking, and mobility through its architecture and use of protocols like SIP, SDP, RTP, and MGCP.
Introduction to SIP(Session Initiation Protocol)William Lee
Session Initiation Protocol (SIP) is a signaling protocol for managing multimedia communication sessions over Internet Protocol (IP) networks. SIP can be used to establish two-party or multiparty sessions that include voice, video, chat, gaming, and other forms of media. The document introduces SIP architecture, message format, and common call flows including registration, basic call setup, call modification, call hold, and three-way conferencing.
One of the most widely observed fraud scenarios is the case of a malicious user detecting the address of a PSTN gateway and accessing that gateway directly. Once the attacker has managed to access the gateway the attacker can start selling telephony minutes through that
gateway.
The ABC SBC establishes a secure border between the VoIP service provider’s core VoIP components–e.g., PSTN gateway, SIP proxy and application servers- and the subscribers. As the border element, the ABC SBC hides the details of the operator’s network from subscribers
and absorbs any attacks and sudden spikes in the subscriber traffic. Further, the mediation features of the ABC SBC shield the operator’s network from malfunctioning user agents and any interoperability issues.
Interaction Dialer is an outbound dialing solution that can increase revenue, decrease labor costs, and optimize calling campaigns through features like predictive dialing, powerful ACD, and regulatory compliance tools. It provides automated dialing capabilities to improve agent productivity by reducing non-productive tasks. The system uses a single software platform to integrate features like a dialer, recording, reporting, and other functions.
Avaya Aura Application Enablement Services (AES)Motty Ben Atia
The document discusses Avaya's Application Enablement Services (AES) platform. AES provides protocols, APIs, and web services that enable development of modular communication applications. It opens up functionality of Communication Manager and adds communication capabilities to desktop tools. AES supports integration of communication and business applications through API connectors and web services.
This document summarizes hosted VoIP services provided by Reignmaker Communications. It defines VoIP and hosted VoIP, explaining that hosted VoIP provides phone system features over the internet rather than traditional phone lines. It outlines key advantages like lower costs, scalability, and built-in disaster recovery. Finally, it introduces Reignmaker's service offerings and positioning as an end-to-end provider with nationwide coverage and exceptional support.
The document describes the SAPEX IP-PBX server. It is an all-in-one embedded IP-PBX platform that integrates registrar, proxy, presence, and voice mail servers. It supports SIP protocol and allows registration of multiple SIP trunks. Key features include back-to-back user agent for call management, dynamic DNS for seamless connectivity, presence and instant messaging, and RADIUS client for call logging and accounting. The system provides various call routing, call management, and telephony features.
The document describes the features and capabilities of the SAPEX IP-PBX server. It is an all-in-one embedded IP-PBX platform that integrates registrar, proxy, presence and voice mail servers. It supports SIP protocol and allows registration of multiple SIP trunks. Key features include seamless connectivity for remote users, dynamic DNS, presence and instant messaging, call detail recording, RADIUS client for authentication and accounting, and various call management functions.
SPARSH VP248 is a high-definition VoIP phone built with superior acoustics and elegant design to provide unsurpassed audio quality and rich user experience.
Based on open-standard SIP protocol, SPARSH VP248 is interoperable with any standard SIP infrastructure such as IP-PBX, SIP Proxies, Softswitches and Stand-alone applications.
SPARSH VP248 is designed for power users, knowledge workers and managers for quick access totheadvance system features and functions. A feature-packed IP phoneenables user to work efficiently with advance call handling capabilities
Matrix Telecom Solutions: SETU VTEP - Fixed VoIP to T1/E1 PRI GatewayMatrix Comsec
Matrix SETU VTEP is a compact and dedicated gateway for VoIP to T1/E1 PRI network offering high-value communication experience to businesses of all size, Service Providers, Call Centers and simple but cost-effective solution for multi-location branch office communication. This intelligently designed gateway incorporates advanced features with multiple connectivity options to connect with a legacy communication system using T1/E1 or PRI signaling. SETU VTEP offers reliable and cost-effective solutions to the changing requirements of the business communication and offer customer value for money.
The document introduces the Vertical Wave IP 2500, an IP PBX system with integrated applications. It discusses key customer considerations for IP phone systems and the evolution from legacy systems to converged platforms. The Vertical Wave provides scalability, flexibility, investment protection, mobility and ease of management through its applications-inside architecture and web-based administration tools.
Advantage2000ucs version 1-2 InternationalNajam Khan
The document describes the features of Advantage 2000 Unified Communication Server (UCS), a communication platform that provides unified communications, collaboration, and integration capabilities. It includes features such as IP PBX, call center software, CRM software, instant messaging, video calling, conferencing, and supports integration with smartphones, softphones, and third-party software. The UCS provides these capabilities to small and medium businesses through an affordable and easy to manage solution.
The document provides an overview of three main voice over IP protocols: SIP, H.323, and MGCP. It describes the architecture and components of SIP and H.323, including user agents, proxies, registrars, and gateways. It also discusses how SIP and H.323 establish communication sessions and handle registration, call setup and teardown. MGCP is mentioned but not described in detail.
The document describes the features of Advantage 2000 Unified Communication Server (UCS), which integrates various telecommunication and collaboration tools. Key features include IP PBX, fax server, instant messaging, call center software, CRM software and their integration. It also lists specifications of UCS servers and IP gateways for digital trunks and analog ports.
The document provides an overview of Avaya IP Office and its capabilities for delivering intelligent communications solutions to small and midsize businesses. Key points include that Avaya IP Office is an award-winning platform with over 4 million users worldwide, offering a variety of endpoints, user interfaces, mobility features, applications like voicemail pro and call recording, as well as management and administration tools. It also supports voice networking between IP Office sites and SIP trunking.
The document summarizes the Allworx phone system, including its key features and capabilities. It was founded in 1998 and acquired by PAETEC in 2007. It provides IP phone systems for small and medium sized businesses with less than 100 users per location. The systems offer features like voicemail, conferencing, call queues, mobility applications, and support for multiple sites. They are designed to provide a full-featured phone system with a low total cost of ownership.
This document provides an overview of Nexge Technologies, which offers VoIP, OTT, SMS, and VAS solutions to over 250 telecom service providers globally. It summarizes Nexge's key competencies including softswitches, billing systems, messaging platforms, conferencing solutions, and mobile apps. It also outlines Nexge's architecture and solutions for areas like class 4 and 5 switching, SMS, conferencing, call recording, lawful intercept, broadcasting, and more.
SIP is a protocol for initiating, managing, and terminating multimedia sessions over the Internet. It allows users to establish multimedia sessions including voice, video, and text. SIP components include user agents, proxy servers, redirect servers, and registrars. SIP uses request methods like INVITE, ACK, BYE and response codes to set up and terminate sessions between user agents. SIP addresses users in a globally reachable way and allows real-time communication between parties.
Matrix ETERNITY is a family of IP-PBXs with Universal Connectivity and Seamless Mobility. The ETERNITY IP-PBX offers built-in gateway capability to connect nearly all telecom interfaces like FXS, FXO, ISDN BRI, ISDN PRI, T1/E1, GSM and 3G.
This document provides instructions for configuring a desktop phone and firewall for use with a 3CX phone system. It discusses supported phone models, provisioning types, configuring BLF and RPS phones, troubleshooting, and using the 3CX web client. It also covers configuring firewall ports, NAT/port settings, SIP trunk providers, inbound and outbound call rules, digital receptionists, and installing 3CX on different operating systems.
iTel switch | Softswitch platform for global Retail, Wholesale, Calling card ...REVE Systems
iTel Switch is a single Softswitch platform for global Retail, Wholesale, Calling card & Call shop business. Being a highly customizable and scalable VoIP Softswitch with integrated billing, it serves as an ideal platform for all the VoIP service providers who want to provide a wide range of VoIP services. iTel VoIP Softswitch has been designed to meet the highest needs of carriers. This VoIP Softswitch also ensures most reliable and cost effective solution that can help VoIP service providers grow as a giant global carrier in VoIP industry. Multilevel reseller support, easy end user interface, integrated billing, intelligent routing and class 4 & 5 Softswitch features are some of the many unique competencies of iTel Switch.
For more details visit:
https://www.revesoft.com/products/itel-voip-softswitch
This document provides an overview of IP and VoIP fundamentals including:
- IP basics such as IP addressing schemes (IPv4 and IPv6), IP ranges, and private IP address ranges.
- VoIP concepts such as devices, codecs, and channels.
- SIP including messages, responses, and call flows for peer-to-peer and proxy calling.
- SIP trunks and their differences from VoIP channels.
- SIP extensions and configuring them on a server.
- VoIP port configuration including LAN, WAN, DNS, and STUN/port forwarding.
- Matrix products that support VoIP including their VoIP channel and trunk capabilities.
- The document discusses Avaya's IP Office product portfolio for small and medium businesses, including its various editions, features, and user productivity solutions.
- Key enhancements in the new Release 5 include increased capacity, simplified licensing, and new features such as SIP trunking and Exchange 2007 integration.
- The document provides information on pricing, discounts, and material codes for IP Office systems, multi-site options, and user licenses.
Talking SIP Sales Presentation that provides an overview of Talking SIP, the industry's award winning and leading Application, Media and Billing Server that powers the top next-generation networks around the world.
Similar to Matrix Telecom Solutions: SAPEX IP-PBX (20)
Leveraging on sophisticated infrastructure and profound knowledge in this domain, we are considered as the reckoned trader and exporter of Video Management Systems
Hybrid Video Surveillance Systems India | Video SurveillanceMatrix Comsec
We manufacture and market one of the best Video Surveillance and Monitoring systems to all of our clients by utilizing state-of-the-art equipment and highly trained dispatchers.
Become a Partner: A Few Good Reasons why it Makes Sense to PARTNER WITH MATRIXMatrix Comsec
The document summarizes an Indian telecom and security solutions company called Matrix that was established in 1991. It has a presence in over 40 countries, 500+ partners, 400+ employees, and has won 25 awards for product engineering and innovation. The company offers over 60 products across several categories including IP-PBXs, time attendance, access control, video surveillance, and more. It has corporate offices in Vadodara, India along with an R&D center and manufacturing unit. Matrix aims to provide its partners with more profit, growth, support, products, and marketing assistance.
Matrix Video Surveillance Solution: SATATYA - The Persistent VisionMatrix Comsec
This document provides an overview of Matrix and its SATATYA video surveillance solution. Matrix is an established Indian manufacturer of security and telecom products with over 350,000 customers in 50+ countries. Its SATATYA solution includes centralized management software, IP and analog cameras, and DVRs. Key benefits of SATATYA include centralized management, remote viewing via web and mobile, integration with other systems, and features like alerts, recording schedules and search. The document outlines the various components, functions, advantages and benefits of the SATATYA video surveillance solution.
Matrix Telecom Solutions: SETU VGB - Fixed VoIP to GSM/3G-ISDN BRI GatewayMatrix Comsec
SETU VGB is an integrated gateway offering connectivity to IP, GSM/3G and ISDN BRI networks on a single platform. The gateway offers access to IP and GSM/3G networks for existing users of ISDN PBX. For an IP-PBX, it provides BRI and GSM/3G trunks connectivity. The gateway offers 8 VoIP channels, 4 GSM/3G SIMs and 2 ISDN BRI ports
Matrix Telecom Solutions: User Terminals IntroductionMatrix Comsec
This document introduces several user terminal products from Matrix including VoIP phones, digital key phones, softphones, and analog telephones. The SPARSH VP330 is highlighted as a VoIP phone with a 4.3-inch color touchscreen display, HD audio quality, and WiFi connectivity. The SPARSH VP248 is a SIP phone that supports 3 SIP accounts and has touch-sensitive programmable keys. Digital key phones like the EON310 and EON48 provide LCD displays, programmable keys, and full-duplex speakerphones. Softphone options like SPARSH MS allow users mobility and integration with contact lists. The presentation provides overviews of features for each terminal type.
Matrix Telecom Solutions: VISION - The Office CommunicatorMatrix Comsec
VISION is a premium feature packed PBX specially designed for modern offices. Micro controller based design and Surface Mount Technology (SMT) based hardware components leads to compact size and robust, reliable and maintenance-free performance. The advance features of the system ensure smooth and efficient call management. VISION provides connectivity to 3 analog trunk lines and 8/9 analog extensions.
Matrix Telecom Solutions: ETERNITY - Building Intercom SolutionMatrix Comsec
To get a good intercom communication solution, one cannot deny the possible security advantages.Matrix ETERNITY is the specialized building intercom system serving dual aspects of communication and security systems for the housing colonies, residential and commercial towers of all sizes. ETERNITY intercom system offers auxiliary ports for direct connectivity with emergency detectors, sensors, paging systems and hooters eliminating the need for additional security systems.
Matrix Telecom Solutions: ETERNITY MEX - The ULSB MK III SwitchMatrix Comsec
The document introduces the ETERNITY MEX, a military exchange switch. It has a compact and ruggedized form factor compliant with military standards. It functions as a local/transit switch using digital PCM-TDM and IP technology with high-density 1024x1024 switching. It has a modular architecture with redundancy and hot-swappable cards. Key features include extensive diagnostic testing, secure remote management, and ruggedization for harsh military environments.
Matrix Telecom Solutions: NAVAN CNX200 - Office-in-a-Box Solution for Small B...Matrix Comsec
NAVAN CNX200 is an all-in-one office solution for small businesses and enterprise branch offices with up to 24 users. It combines the functionalities of IP-PBX, Data router, Wi-Fi access point, VoIP-GSM gateway, VPN and Firewall Security in a compact and converged platform. A true office-in-a-box, CNX200 innovates the way small businesses communicate and manage infrastructure, so that they can increase productivity, lower costs and enhance collaboration with customers and suppliers.
Matrix Security Solutions: SATATYA HVR - Hybrid Video RecordersMatrix Comsec
Prompt video surveillance goes a long way in enhancing the security, productivity and safety of any organization. Most of them have to compromise between cost of the solution and the effectiveness it offers, i.e. between analog cameras and IP cameras. The SATATYA HVR is an instant solution which utilizes the cost benefits of analog solution and security enhancements of IP solution. Moreover, SATATYA Centralized Management Software (CMS), HVR Client and Mobile Viewer provide software solutions for centralized and remote management, recording and monitoring of the entire organization.
Matrix Security Solutions: SATATYA NVR - Network Video RecorderMatrix Comsec
SATATYA NVR400 is a 4, 8, 12, or 16 channel network video recorder that provides centralized surveillance management. Key features include H.264 and MPEG-4 video compression, remote accessibility via web and mobile clients, storage backup to USB/NAS drives, RAID support, and support for ONVIF and other IP camera brands. The NVR400 has comprehensive alarm handling including email/SMS alerts and supports up to 4TB of storage.
Various establishments like corporate, hostels, schools, hospitals often support their own canteen. Till date the entire canteen process from controlling long queues to food selection to payment is manual and subject to inaccuracies. Matrix e-Canteen management module is a completely automated solution starting from placing an order to delivery including the payment which helps overcome these issues. It allows item tracking, secure and speedy transaction; prevents wastage of food and error in accounting. It helps management to handle the users smoothly thus reducing the waiting time for the users.
Matrix Roster Management is about planning and best utilization of workforce and managing the cost of manpower efficiently. It is designed to simplify the process of defining and managing employee duty rosters. It also provides a complete roster plan with the list of employees, their work time and area of work. Moreover, export data field to XML file makes third party payroll integration easy. Matrix Roster Management saves time and cost significantly by placing right people with the right skill at the right job.
Matrix Security Solutions: SATATYA DVR - Digital Video RecordersMatrix Comsec
This document introduces the SATATYA digital video recorder (DVR) system from Matrix. It provides an overview of DVRs and their functions such as playback, storage, compression, and recording. It then describes the SATATYA DVR hardware and software solution, highlighting features like H.264 compression, pentaplex functions, advanced recording and search options, remote access and management, and integrated security cameras. The document aims to demonstrate how SATATYA DVR provides a comprehensive surveillance solution for small, medium, and large businesses.
Matrix Security Solutions: COSEC - Access Control and Time-AttendanceMatrix Comsec
Matrix COSEC Time-Attendance is a perfect solution for any type of organizations. It offers superlative range of flexible functions like Shifts and Schedules, Late-In and Early-Out, Overtime, Comp-OFF, Absenteeism, Multiple Organizations, Leave Management, Past Adjustments, etc. Moreover system can generate 100+ reports and charts for maintaining well organized employee database and for easy interpretation. Employee Self Service portal is a powerful software tool for employees and their reporting officer to plan shift schedules, request and approve leaves, view attendance record, manually correct timing etc.
Matrix Telecom Solutions: SETU VGFX - Fixed VoIP to GSM/3G-FXO-FXS Voice Gat...Matrix Comsec
Matrix presents SETU VGFX- The Single-box Gateway solution, offering seamless connectivity between VoIP, GSM and POTS (FXO and FXS) networks. SETU VGFX supports flexible and intelligent call routing options to ensure that communication always happens through the most cost effective network.
Matrix Telecom Solutions: SETU VFXTH - Fixed VoIP to FXO-FXS GatewaysMatrix Comsec
Matrix presents SETU VFXTH-The multi-channel SIP gateway offering seamless connectivity between VoIP and PSTN networks through multiple FXS and FXO ports. Matrix SETU VFXTH offers universal and transparent call routing irrespective of type of ports – VoIP-FXS, VoIP-FXO and FXS-FXO. Its superior call and signal processing capabilities ensure unrestricted flow of multiple calls with higher speed and better speech quality.
Matrix Telecom Solutions: SETU VFX - Fixed VoIP to FXO-FXS GatewaysMatrix Comsec
Matrix SETU VFX is designed to meet these requirements of converting VoIP network to traditional telephony interfaces and vice-versa. It handles all the complexities of VoIP technology internally and provides simple telephone interfaces to make and receive calls.
Building Production Ready Search Pipelines with Spark and MilvusZilliz
Spark is the widely used ETL tool for processing, indexing and ingesting data to serving stack for search. Milvus is the production-ready open-source vector database. In this talk we will show how to use Spark to process unstructured data to extract vector representations, and push the vectors to Milvus vector database for search serving.
Sudheer Mechineni, Head of Application Frameworks, Standard Chartered Bank
Discover how Standard Chartered Bank harnessed the power of Neo4j to transform complex data access challenges into a dynamic, scalable graph database solution. This keynote will cover their journey from initial adoption to deploying a fully automated, enterprise-grade causal cluster, highlighting key strategies for modelling organisational changes and ensuring robust disaster recovery. Learn how these innovations have not only enhanced Standard Chartered Bank’s data infrastructure but also positioned them as pioneers in the banking sector’s adoption of graph technology.
Let's Integrate MuleSoft RPA, COMPOSER, APM with AWS IDP along with Slackshyamraj55
Discover the seamless integration of RPA (Robotic Process Automation), COMPOSER, and APM with AWS IDP enhanced with Slack notifications. Explore how these technologies converge to streamline workflows, optimize performance, and ensure secure access, all while leveraging the power of AWS IDP and real-time communication via Slack notifications.
“An Outlook of the Ongoing and Future Relationship between Blockchain Technologies and Process-aware Information Systems.” Invited talk at the joint workshop on Blockchain for Information Systems (BC4IS) and Blockchain for Trusted Data Sharing (B4TDS), co-located with with the 36th International Conference on Advanced Information Systems Engineering (CAiSE), 3 June 2024, Limassol, Cyprus.
Dr. Sean Tan, Head of Data Science, Changi Airport Group
Discover how Changi Airport Group (CAG) leverages graph technologies and generative AI to revolutionize their search capabilities. This session delves into the unique search needs of CAG’s diverse passengers and customers, showcasing how graph data structures enhance the accuracy and relevance of AI-generated search results, mitigating the risk of “hallucinations” and improving the overall customer journey.
For the full video of this presentation, please visit: https://www.edge-ai-vision.com/2024/06/building-and-scaling-ai-applications-with-the-nx-ai-manager-a-presentation-from-network-optix/
Robin van Emden, Senior Director of Data Science at Network Optix, presents the “Building and Scaling AI Applications with the Nx AI Manager,” tutorial at the May 2024 Embedded Vision Summit.
In this presentation, van Emden covers the basics of scaling edge AI solutions using the Nx tool kit. He emphasizes the process of developing AI models and deploying them globally. He also showcases the conversion of AI models and the creation of effective edge AI pipelines, with a focus on pre-processing, model conversion, selecting the appropriate inference engine for the target hardware and post-processing.
van Emden shows how Nx can simplify the developer’s life and facilitate a rapid transition from concept to production-ready applications.He provides valuable insights into developing scalable and efficient edge AI solutions, with a strong focus on practical implementation.
HCL Notes und Domino Lizenzkostenreduzierung in der Welt von DLAUpanagenda
Webinar Recording: https://www.panagenda.com/webinars/hcl-notes-und-domino-lizenzkostenreduzierung-in-der-welt-von-dlau/
DLAU und die Lizenzen nach dem CCB- und CCX-Modell sind für viele in der HCL-Community seit letztem Jahr ein heißes Thema. Als Notes- oder Domino-Kunde haben Sie vielleicht mit unerwartet hohen Benutzerzahlen und Lizenzgebühren zu kämpfen. Sie fragen sich vielleicht, wie diese neue Art der Lizenzierung funktioniert und welchen Nutzen sie Ihnen bringt. Vor allem wollen Sie sicherlich Ihr Budget einhalten und Kosten sparen, wo immer möglich. Das verstehen wir und wir möchten Ihnen dabei helfen!
Wir erklären Ihnen, wie Sie häufige Konfigurationsprobleme lösen können, die dazu führen können, dass mehr Benutzer gezählt werden als nötig, und wie Sie überflüssige oder ungenutzte Konten identifizieren und entfernen können, um Geld zu sparen. Es gibt auch einige Ansätze, die zu unnötigen Ausgaben führen können, z. B. wenn ein Personendokument anstelle eines Mail-Ins für geteilte Mailboxen verwendet wird. Wir zeigen Ihnen solche Fälle und deren Lösungen. Und natürlich erklären wir Ihnen das neue Lizenzmodell.
Nehmen Sie an diesem Webinar teil, bei dem HCL-Ambassador Marc Thomas und Gastredner Franz Walder Ihnen diese neue Welt näherbringen. Es vermittelt Ihnen die Tools und das Know-how, um den Überblick zu bewahren. Sie werden in der Lage sein, Ihre Kosten durch eine optimierte Domino-Konfiguration zu reduzieren und auch in Zukunft gering zu halten.
Diese Themen werden behandelt
- Reduzierung der Lizenzkosten durch Auffinden und Beheben von Fehlkonfigurationen und überflüssigen Konten
- Wie funktionieren CCB- und CCX-Lizenzen wirklich?
- Verstehen des DLAU-Tools und wie man es am besten nutzt
- Tipps für häufige Problembereiche, wie z. B. Team-Postfächer, Funktions-/Testbenutzer usw.
- Praxisbeispiele und Best Practices zum sofortigen Umsetzen
HCL Notes and Domino License Cost Reduction in the World of DLAUpanagenda
Webinar Recording: https://www.panagenda.com/webinars/hcl-notes-and-domino-license-cost-reduction-in-the-world-of-dlau/
The introduction of DLAU and the CCB & CCX licensing model caused quite a stir in the HCL community. As a Notes and Domino customer, you may have faced challenges with unexpected user counts and license costs. You probably have questions on how this new licensing approach works and how to benefit from it. Most importantly, you likely have budget constraints and want to save money where possible. Don’t worry, we can help with all of this!
We’ll show you how to fix common misconfigurations that cause higher-than-expected user counts, and how to identify accounts which you can deactivate to save money. There are also frequent patterns that can cause unnecessary cost, like using a person document instead of a mail-in for shared mailboxes. We’ll provide examples and solutions for those as well. And naturally we’ll explain the new licensing model.
Join HCL Ambassador Marc Thomas in this webinar with a special guest appearance from Franz Walder. It will give you the tools and know-how to stay on top of what is going on with Domino licensing. You will be able lower your cost through an optimized configuration and keep it low going forward.
These topics will be covered
- Reducing license cost by finding and fixing misconfigurations and superfluous accounts
- How do CCB and CCX licenses really work?
- Understanding the DLAU tool and how to best utilize it
- Tips for common problem areas, like team mailboxes, functional/test users, etc
- Practical examples and best practices to implement right away
GraphSummit Singapore | The Art of the Possible with Graph - Q2 2024Neo4j
Neha Bajwa, Vice President of Product Marketing, Neo4j
Join us as we explore breakthrough innovations enabled by interconnected data and AI. Discover firsthand how organizations use relationships in data to uncover contextual insights and solve our most pressing challenges – from optimizing supply chains, detecting fraud, and improving customer experiences to accelerating drug discoveries.
AI 101: An Introduction to the Basics and Impact of Artificial IntelligenceIndexBug
Imagine a world where machines not only perform tasks but also learn, adapt, and make decisions. This is the promise of Artificial Intelligence (AI), a technology that's not just enhancing our lives but revolutionizing entire industries.
Essentials of Automations: The Art of Triggers and Actions in FMESafe Software
In this second installment of our Essentials of Automations webinar series, we’ll explore the landscape of triggers and actions, guiding you through the nuances of authoring and adapting workspaces for seamless automations. Gain an understanding of the full spectrum of triggers and actions available in FME, empowering you to enhance your workspaces for efficient automation.
We’ll kick things off by showcasing the most commonly used event-based triggers, introducing you to various automation workflows like manual triggers, schedules, directory watchers, and more. Plus, see how these elements play out in real scenarios.
Whether you’re tweaking your current setup or building from the ground up, this session will arm you with the tools and insights needed to transform your FME usage into a powerhouse of productivity. Join us to discover effective strategies that simplify complex processes, enhancing your productivity and transforming your data management practices with FME. Let’s turn complexity into clarity and make your workspaces work wonders!
Removing Uninteresting Bytes in Software FuzzingAftab Hussain
Imagine a world where software fuzzing, the process of mutating bytes in test seeds to uncover hidden and erroneous program behaviors, becomes faster and more effective. A lot depends on the initial seeds, which can significantly dictate the trajectory of a fuzzing campaign, particularly in terms of how long it takes to uncover interesting behaviour in your code. We introduce DIAR, a technique designed to speedup fuzzing campaigns by pinpointing and eliminating those uninteresting bytes in the seeds. Picture this: instead of wasting valuable resources on meaningless mutations in large, bloated seeds, DIAR removes the unnecessary bytes, streamlining the entire process.
In this work, we equipped AFL, a popular fuzzer, with DIAR and examined two critical Linux libraries -- Libxml's xmllint, a tool for parsing xml documents, and Binutil's readelf, an essential debugging and security analysis command-line tool used to display detailed information about ELF (Executable and Linkable Format). Our preliminary results show that AFL+DIAR does not only discover new paths more quickly but also achieves higher coverage overall. This work thus showcases how starting with lean and optimized seeds can lead to faster, more comprehensive fuzzing campaigns -- and DIAR helps you find such seeds.
- These are slides of the talk given at IEEE International Conference on Software Testing Verification and Validation Workshop, ICSTW 2022.
Maruthi Prithivirajan, Head of ASEAN & IN Solution Architecture, Neo4j
Get an inside look at the latest Neo4j innovations that enable relationship-driven intelligence at scale. Learn more about the newest cloud integrations and product enhancements that make Neo4j an essential choice for developers building apps with interconnected data and generative AI.
Why You Should Replace Windows 11 with Nitrux Linux 3.5.0 for enhanced perfor...SOFTTECHHUB
The choice of an operating system plays a pivotal role in shaping our computing experience. For decades, Microsoft's Windows has dominated the market, offering a familiar and widely adopted platform for personal and professional use. However, as technological advancements continue to push the boundaries of innovation, alternative operating systems have emerged, challenging the status quo and offering users a fresh perspective on computing.
One such alternative that has garnered significant attention and acclaim is Nitrux Linux 3.5.0, a sleek, powerful, and user-friendly Linux distribution that promises to redefine the way we interact with our devices. With its focus on performance, security, and customization, Nitrux Linux presents a compelling case for those seeking to break free from the constraints of proprietary software and embrace the freedom and flexibility of open-source computing.
TrustArc Webinar - 2024 Global Privacy SurveyTrustArc
How does your privacy program stack up against your peers? What challenges are privacy teams tackling and prioritizing in 2024?
In the fifth annual Global Privacy Benchmarks Survey, we asked over 1,800 global privacy professionals and business executives to share their perspectives on the current state of privacy inside and outside of their organizations. This year’s report focused on emerging areas of importance for privacy and compliance professionals, including considerations and implications of Artificial Intelligence (AI) technologies, building brand trust, and different approaches for achieving higher privacy competence scores.
See how organizational priorities and strategic approaches to data security and privacy are evolving around the globe.
This webinar will review:
- The top 10 privacy insights from the fifth annual Global Privacy Benchmarks Survey
- The top challenges for privacy leaders, practitioners, and organizations in 2024
- Key themes to consider in developing and maintaining your privacy program
5. Server Specifications
Open-Architecture, Supports SIP v2 Protocol
Embedded Registrar, Proxy, Voice Mail and Presence Servers
Back-to-Back User Agent (B2BUA)
Registration of Multiple SIP Trunks
Seamless User Connectivity (NAT and STUN Support)
Dynamic DNS (DDNS) Support
Presence and Instant Messaging
6. Open Architecture
SAPEX Employs Open-Standard SIP Protocol to Establish Calls
Over IP Network
Full Interoperability With Any Third-party SIP Equipment Such as IP
Phones, Softphones, Gateways and Proxy Servers
7. Open Architecture
SIP is a Open-Standard Signaling Protocol for Establishing
Communication Session over Internet
Voice, Video or Instant Messaging Sessions
SIP Architecture
SIP Servers and SIP Endpoints (User Agents/Terminals)
Varied Options of Communication Terminals
PC, PDA, Cell Phone With SIP Client, IP Phone, Softphones
8. SIP Servers Types
Registrar Server
Authenticates and Registers When User Comes Online
Proxy Server
Processes Session Requests and Responses of a SIP Call
Redirect Server
Redirects Calls to an Active SIP User Terminal
Presence Server
Stores and Distribute Users’ Presence Information
9. The All-in-One IP-PBX Server
Presence
Server
Proxy
Server
Registrar
Server
Voice Mail
Server
Embedded Registrar, Proxy, Presence and Voice Mail Server
10. Back-to-Back User Agent (B2BUA)
Basic Call Services like Call Forwarding, Call Transfer
Necessitates Call Management and Tracking for Entire Call
Duration
SIP Server With B2BUA Becomes an Active Participant in a SIP
Call, Enabling Many Advanced Services in Addition to Basic
Telephony Services
Applications Such as Billing Require Call State Monitoring
Facilitates Centralized Call Management
11. Multiple SIP Trunk Support
SAPEX Supports Registration With Multiple SIP Trunks
Registration With a Maximum of 10 SIP Servers is Supported
While Making an Outgoing Call, System Selects a SIP Trunk as per
the Call Routing Algorithms
12. Seamless User Connectivity
A SIP User Located Over Public/Private Network Can be
Registered Easily With SAPEX
A Remote User’s Connectivity is Maintained Even When Behind a
NAT or Firewall
The Embedded Dynamic DNS Client Ensures Round-the-Clock
User Connectivity
A User Can Have Multiple Contact Points (Terminals), Mapped to a
Common User Identity
A User Can be Called on Any/All of the Active Terminal at a Given
Instant
13. Dynamic DNS (DDNS)
Automates the Discovery and Registration of the Server’s IP Address on
the Public Network
Fluctuating Dynamic IP of the Server is Mapped to a Unique Domain
Name (DNS)
Remote Administrator and IP-PBX Clients Can Connect to the Server
Using the Non-Altering DNS
Benefit:
Prevents Reconfiguring of Systems Every Time a Network Infrastructure Changes
Useful When Remotely Accessing the IP-PBX Connected to Ones Home or Office IP
Connectivity, Usually Configured for a Dynamic IP
14. Presence and IM
Before an Actual Conversation Begins, Presence Determines: :
The Availability of a User (Such As Online, Offline and Others)
His Willingness to Participate in a Communication Session (Busy, Available
on Phone, Out of Office And Others)
His Preferred Mode of Communication (Call or Instant Messaging)
The Presence Server Maintains and Distributes Presence Information of
Users Registered With the IP-PBX
Instant Messaging (IM) is a Popular Mode of Real-time Communication
Knowing the Availability Status of Users, Reach a Right Contact, In Right
Time and on the Right Terminal
15. Management Features
Enhanced Administrative Ease
World-Wide Portability of Extensions
Busy Lamp Field (BLF)
Call Detail Records (CDR)
RADIUS Client
Call Management
16. Enhanced Administrative Ease
Route Calls Also Over IP Network, Besides Data
Embedded Web Server and Web Based Programming GUI
With a Few Clicks From the Intuitive GUI (Jeeves) Itself:
An Administrator Can Remotely Program and Configure the System
Administrator Can Monitor and Manage User Registration, Feature
Access in Real-time
A New User Can be Added
User’s Calling Rights Can be Defined
User Groups Can be Formed
Voice Mail Can be Allocated
17. Extension Portability
In IP Telephony, a VoIP Call is Established via IP Network
Instead of a Phone Number, An IP User is Identified by SIP URI
An IP User is Free to Port his Extension Anywhere on the IP
Network
Users Can Establish Calls Retaining the Same Contact Credentials,
though Registering From Varied Locations
Unlike Traditional Telephony System, SAPEX Does Not Bind
the User to a Fixed Location
18. Busy Lamp Field (BLF)
A Busy Lamp Field is An Array of Extension Status Lamps
An Extension Can Bear Different Statuses Such as:
Busy, Ringing or Idle
If a User’s Class of Service (Cos) is Provisioned, An Extension
User Can Monitor the Status of Another Extension
Based on Extension Status, the Operator Can Decide to Either
Transfer a Call Directly or Else Pick Up a Call Incase Called
Extension is Busy
19. Call Detail Records (CDR)
Call Details are Stored in the System’s CDR Buffer
The Call Log is Stored for Different Types of Calls
Internal Calls Made Between System Users
External Calls Made Between System User and External User
The IP-PBX Can be Configured to Send CDR Text-File as Email
Attachments
20. Call Detail Records (CDR)
CDR Report Details:
Calling/Called Number
SIP Trunk Used for External Calls
Date and Time of Call
Call Duration
Call Termination Cause
21. RADIUS Client
SAPEX Logs the Details of Calls in CDR Files
These CDR Files Contain Essential Information to Account a User
for the Services Utilized by Him
These CDR Files are Therefore Requiring a Safe and Longer
Storage
The Embedded RADIUS Client Facilitates Efficient Call Logging to a
Remote Server/Database
22. RADIUS Client
RADIUS : Remote Authentication Dial In User Service
Enables the IP-PBX to Send the CDR Files to a Centrally Managed
Remote Server Called RADIUS Server/Database
It Employs an Authentication, Authorization And Accounting ClientServer Protocol (AAA Client-Server Protocol)
Further Integration With a Billing Server, Can Benefit the Service
Providers in Accurate Billing
23. AAA Procedure
Authentication
Validating a User
Authorization
Defining Permissible Services for a User
Accounting
Keep a Track of Resources Utilization by a User for the Purpose of
Billing and Monitoring
24. RADIUS Client-Basic Operation
A
Calling Party
IP
A
A
RADIUS
Server
SAPEX
IP Phone
User
Called Party
Authentication
Authorization
Call Established
Accounting
Billing
Server
25. Call Management
Incoming Call Management Features:
Anonymous Call Rejection
Auto Attendant
Caller-ID Based Routing
Do-Not-Disturb
DDI Routing
Time Tables
User Groups
26. Call Management
Outgoing Call Management Features:
Allowed/Denied Number List
Automatic Number Translation
Dial Plans
Emergency Number Dialing
Peer-to-Peer Calls
Reverse DDI
27. Call Management
Telephony Services:
Call Forward
Call Hold
Call Pickup (Selective and Group)
Call Park (Personal Orbit), Call Retrieve (Personal and Global Orbit)
Call Transfer (Blind and Attended)
CLIR
Conference
28. Anonymous Call
Rejection
An Incoming Call Without Caller Line Identification (CLI) Number is
Termed as Anonymous Call
Instead of Number, the term “Anonymous” is Displayed on the
Screen
Offers the Flexibility to Directly Reject Anonymous Calls or Route
Such Calls to a Specific Extension
29. Auto Attendant
The Auto Attendant Informs the Caller of the Way to Reach His
Ultimate Destination
Customized Welcome and Guiding Prompts as per Time of the Day,
Music-on-Hold Can be Played to a Caller
With the Automated Attendant, a Caller Can Find-his-way to:
Reach to a Desired Extension
Retrieve Information
Leave Back a Message in the Mail Box of the Called Extension
30. Voice Mail Features
Configurable Voice Mail Server Size
Individual Voice Mail for Each User
Configurable Mailbox Size
Customizable Greetings
Voice Mail Notification by E-mail Using SMTP
31. Caller-ID Based Routing
Calls Get Routed to Pre-defined Extensions as per CLI of Calling Party
A Calling Party Number Table Can be Programmed
The CLI of Calling Party is Tallied With the Table Entries
Calls are then Routed to Defined Stations as per Routing Groups
Depending on the Time, Call Can be Routed to Different Destinations
For example:
Calls Important to Business May be Directed to Higher Authorities
Calls With Specific CLI May be Directed to Particular Departments
While Calls From Anonymous Numbers May be Directed to the Customer
Support Teams
32. Do-Not-Disturb
Do-Not -Disturb (DND) Feature Offers Users With the Flexibility of Not
Receiving Calls for Particular Time Period
Outgoing Calls Can be Made When DND is Enabled
33. DDI Routing
A Call Landing on SIP Trunk Can be Directly Routed to an Extension
as per the DDI Numbering
The DDI Facility Should be Activated on the SIP Trunk by the SIP
Service Provider
Unlike Traditional Telephony Services, IP Telephony Does Not Bind
a Number to its Geographical Location
Calls are Placed Over Internet and Numbers are Mapped to IP
Addresses, Which May be Anywhere on the Internet
An IP Extension Can Always be Called Irrespective of Its Current
Location
34. Time Table
Route Incoming Calls as per ‘Time of the Day’ (Time Zone)
Defined Schedule for a Day is Called Time Table
A Timetable Divides Entire Day in to Different Time Zones
4 Such Timetables Can be Defined
Provides Flexibility to Receive Ones Calls on Different Terminals as
per the Time
35. User Groups
Multiple Extensions Can be Clustered as One Group
Defining User Groups, Calls Can be Distributed Between the Group
Members
The Group Members Can be Located at Different Geographic
Locations
On Reception of a Call, the Extensions Ring as per the Assigned
Priorities
Thus Ensuring No Call Remain Unanswered
36. Allowed and Denied Numbers
Allow/Deny Dialing of Specific Numbers
Avoids Misuse and Restricts Unproductive Calls
16 Such Allowed and Denied Number Lists Can be Programmed
37. Automatic Number Translation
SAPEX Supports Registration of Multiple SIP Trunks
These Trunks Can be Availed From Single or Multiple ITSPs
While Placing a Call, a Caller is Not Conscious of the Routing Logics
Defined and the SIP Account in Use
SAPEX Itself Modifies the Dialed Number or Part Thereof, Such that it
Matches With the Numbering Plan that Is Understood by the ITSP
For Example:
If a User Dials 223344 to Call www.abc.com
SAPEX Adds the Appropriate Access Code "*777" Specified by the ITSP and
Dials-Out the Number “*777223344” Instead of 223344
38. Dial Plans
SAPEX Supports Multiple SIP Trunk Registrations
Registration With Maximum of 10 SIP Servers is Supported
Calling Rates Differ on the Basis of Area of Call, Service Provider,
Call Time, etc
A Dial Plan Allows a User to Place Call through the Most Costeffective SIP Trunk, as Per a Defined Call Routing Logic
Each User Can be Assigned Multiple Dial Plans
The Dial Plans May be Same For All Users or May Differ Individually
39. Emergency Number Dialing
Emergency Calls are Not Subjected to Outgoing Call Management
Rules
This Reduces Any Latency While Placing Emergency Calls
SIP Trunk Can be Specified for Such Calls
Four Such Numbers Can be Programmed
40. Peer-to-Peer Calling
Calls Can be Placed Between Two SIP Devices, Without Going
through a Proxy Server
Fixed IP Addresses of Various SIP Devices Can be Programmed in a
Peer-to-Peer Table
500 Such Entries Can be Programmed
43. Reverse DDI
This Feature Activates Carting of DDI Numbers as Caller ID When a
User Makes a Call via SIP Trunk
When a Call is Received From the Called Party, the Call Can be
Directly Routed to the DDI Number Which Placed the Call
Eliminates the Hassle of Searching the DDI User Who Made the Call
Saves Time and Enhances Productivity
44. FAX Homing
FAX Homing Allows a SIP Trunk to be Used for Both:
Voice Calls and to Receive FAX
System Detects FAX Tone on SIP Trunk Only When Call is Answered by the
Auto Attendant
When FAX Tone is Detected, System Routes Call to the Extension Where
FAX Machine is Connected
SIP Trunk
45. Voice Transcoding
There is Diversity Among Available SIP Endpoints (Terminals) and
their Capabilities
Audio Transcoding Helps to Establish Communication Between SIP
Devices With Diverse Audio(Codec) Specifications
Reduces the Ratio of Dropped Calls
49. SAPEX with an ATA
PHONE And PC
Connectivity
SIP PROXY
Remote Client 1
Host
IP NETWORK
PSTN
Remote Client 2
SAPEX
PBX
Shared IP Line
For PBX Extensions
50. SAPEX with VoIP-FXS Gateway
SIP PROXY
IP NETWORK
Standard Phone
Users
SETU VFX
VoIP-FXS Gateway
SAPEX
57. Hardware Specifications
PORTS
WAN Port 1(RJ45, 10/100/1000 Base T)
LAN Port 1(RJ45, 10/100/1000 Base T)
USB Port 1 (Internal)
DC Jack
1 (DC Power Jack)
LED Indications
1 for Power Status and 1 for SIP Trunk Status
Power Supply
External Adaptor 5V DC / 70A
Power Consumption
20 W (Maximum)
Dimensions (WxHxD)
230mmX55mmX163mm (9.06”X2.17”X6.42”)
Installation
Wall Mount and Table-Top
59. Matrix VoIP Product Range
ETERNITY IP-PBX
The IP-PBX with Universal Connectivity and Seamless Mobility
SAPEX
All-in-One Embedded IP-PBX Server
VYOM CCX
High-Density SIP Gateway
ETERNITY
The Universal Telephony Gateway
SETU VGFX
Multi-Port SIP based VoIP to GSM-FXO-FXS Gateway
SETU VFXTH
Multi-Port VoIP to FXO-FXS Gateway
SETU VFX
SIP based VoIP Gateway with 4/8 FXS Ports, 1 FXO (PSTN Pass-Through) and 1 Ethernet Port
SETU ATA211G
SIP based Analog Telephone Adaptor with 1 GSM, 1 FXS Port and 2 Ethernet Ports
SETU ATA211
SIP based Analog Telephone Adaptor with 1 FXO, 1 FXS Port and 2 Ethernet Ports
SETU ATA2S
SIP based Analog Telephone Adaptor with 2 FXS Ports and 2 Ethernet Port
SETU ATA1S
SIP based Analog Telephone Adaptor with 1 FXS Port and 2 Ethernet Ports
SETU VP248PE
Executive IP-Phone with 6 Lines x 24 Characters LCD Display and PoE
SETU VP248SE
Executive IP-Phone with 2 Lines x 24 Characters LCD Display and PoE
SETU VP248P
Executive IP-Phone with 6 Lines x 24 Characters LCD Display
SETU VP248S
Executive IP-Phone with 2 Lines x 24 Characters LCD Display
61. Type of Presentation: Product Presentation
Number of Slides: 61
Revised On:15th July, 2010
Version-Release Number: V1R1
For Further Information Please Contact:
Email ID: mt.pbx@matrixcomsec.com
Mobile: +91 9662544401
Visit us at www.MatrixComSec.com
Editor's Notes
Registrar Server When users come online, they need to make sure that others are aware that they’re available
to take and make calls The Registrar authenticates and registers users when they come online, and then stores
information on the users’ logical identities and the devices that they can use for communications The devices are
identified by their URIs
Location Service As users roam, the network needs to be continually aware of their locations The location
service is a database that keeps track of users and their locations The location service gets its input from the
registrar server and provides key information to the proxy and redirect servers A SIP proxy or redirect server uses this information to obtain the mapping from logical SIP addresses to physical SIP addresses, so that communication sessions can be properly established
and maintained
Redirect Server If users are not in their home domains, sessions bound for them needs to be redirected to them
The redirect server maps a SIP request destined for a user to the URL of the device “closest” to the user
For example, if a call is destined for eileendover@companycom and the user is on the road, the company’s
redirect server may reply to the caller’s user agent (or to the requesting proxy server) with the
contact address of the user’s cell phone, so that the incoming call can be redirected to the cell phone
Proxy Server A proxy server takes SIP requests, processes them, and passes them downstream while
sending responses upstream to other SIP servers or devices A proxy server may act as both a server and
a client, and can modify a SIP request before passing it along A proxy is involved only in the setup and teardown
of a communication session After user agents establish a session, communications occur directly
between the parties
Presence Server In order for users to see the presence of their buddies to improve communication, they often
refer to a presence server Presence servers accept, store, and distribute presence information The presence
server has two distinct sets of clients:
• Presentities (producers of information) provide presence information about themselves to the
server to be stored and distributed
• Watchers (consumers of information) receive presence information from the server Watchers
can subscribe to certain users, much like IM users choose which buddies to add to their list
Presence Means:
User status (that is, online or offline)
User availability (such as available, busy, on the phone, or out to lunch)
User’s desired contact means (such as instant messaging, desk phone, cell phone, pager, and so on)
If a user wants to refer older call details, user will have to take CDR print out very often as the records get overwritten in case the buffer fills to its maximum capacity In such cases, RADIUS enables the system to send the call details/ logs to a centrally managed remote server called RADIUS server/Database
FAX Homing Disabled:
Dial 2630666
Wait for FAX Tone on FAX Machine
Press “Start” to Send FAX
FAX Call is Routed to User 2001
FAX Homing Enabled:
Dial Number 2630666
You get FAX prompt for pressing start button on FAX machine to send FAX
Wait to time you listen the FAX Tone
Press start on your FAX machine
FAX Call is routed to 2001